Voice over Internet protocol (VoIP) systems manage the delivery of voice information over the Internet. VoIP involves sending voice information in digital form in discrete packets rather than using the traditional circuit-committed protocols of the public switched telephone network (PSTN). VoIP is also referred to as IP Telephony, Internet telephony, Broadband telephony, Broadband Phone, and Voice over Broadband. A major advantage of using VoIP is that it avoids the tolls charged by ordinary telephone service providers. As such, VoIP systems are becoming ever more common within enterprises.
A VoIP call typically involves a signaling session and a media session. The signaling can be accomplished using various protocols such as Session Initiation Protocol (SIP), H.323 Protocol, or any other suitable signaling protocols. SIP is an application-layer control (signaling) protocol that is used for creating, modifying, and terminating sessions with one or more participants. These sessions can include Internet telephone calls, multimedia distribution, and multimedia conferences. SIP clients use Transmission Control Protocol (TCP) or User Datagram Protocol (UDP) to connect to SIP servers and other SIP endpoints. H.323 defines the protocols that provide audio-visual communication sessions on any packet network; and is commonly used in VoIP and IP-based videoconferencing.
Media streams are sent using the Real-time Transport Protocol (RTP). RTP helps to ensure that packets get delivered in a timely way. Media streams also involve UDF packets, and are transmitted at regular intervals. Media streams are typically encoded using a speech compression algorithm.
Typically, a call agent handles VoIP call routing for VoIP clients. The call agent typically makes a VoIP call using a destination telephone number. This number can be associated with a client on the same call agent, in which case the call is sent directly to that client. Or, the number might be associated with a client associated with a different agent within the same enterprise. In that case, the call agent sends the call to that agent, using configured rules that define how to route the call. When users within the enterprise communicate with users outside of the enterprise, the call is terminated on a PSTN gateway and routed off to the PSTN. This, however, eliminates many of the benefits of VoIP.
Service providers are now beginning to offer “SIP Trunk” services, whereby an enterprise can connect their calls via a SIP link to the service provider instead of through an enterprise gateway. This technique has many of the same limitations as a direct gateway interconnect, since the service provider typically routes the call to a gateway. Furthermore, SIP trunks are not likely to be much cheaper than time-division multiplexing (TDM) trunks, because there is little business incentive for a service provider to make them more cost effective.
Within enterprises or groups of enterprises, two call agents may connect to each other directly over IP, without requiring an intermediate service provider for voice services. One solution for accomplishing this is to statically configure direct SIP or H.323 trunks between call agent or call manager instances in different enterprises. While this may work for small-scale close knit communities, it becomes very burdensome for even a few dozen interconnected sites and limits its advantages to VoIP calls within the community. Ideally, VoIP should be as easily interconnected as email—any enterprise should just be able to connect to any other enterprise, without configuration.